Data arranging method and medium for data recording or transfer, and signal processing apparatus for the method and medium

ABSTRACT

A data arranging method for linear PCM data, which allows both low-class and high-class machines to easily perform a reproduction process and can cope with multiple channels. Data having a structure in which each sample data of 20 bits or 24 bits of individual channels is separated to a main word consisting of 16 bits and an extra word consisting of 4 or 8 bits, a collection of 2n-th main words of the individual channels is arranged, a collection of (2n+1)-th main words of the individual channels is then arranged, a collection of 2n-th extra words of the individual channels is then arranged, and a collection of (2n+1)-th extra words of the individual channels is then arranged, is recorded on a recording medium or transferred.

BACKGROUND OF THE INVENTION

The present invention relates to a data arranging method and a mediumfor recording or transferring data or the like to be recorded on adigital video disk and a digital audio disk, and a signal processingapparatus for processing the data.

Recently, digital video disks have been developed as optical disks inaddition to conventional compact disks (hereinafter referred to as"CDs") for audio usage, and players for such digital video disks havealso been developed. In particular, the digital video disks include akind which is about the same size (12 cm in diameter) as theconventional CDs and is designed such that about two hours of pictureinformation can be recorded on and reproduced from that disk. For such adigital video disk, there is a format which allows voices or music ineight different languages and superimposition information in thirty-twodifferent languages to be recorded on the same disk in addition topicture information.

Again, digital video disks which can record voices or music in multiplelanguages in addition to main picture information and are the same sizeas the conventional CDs have been developed.

If such digital video disks become available on the market, naturally,it would be a natural demand to reproduce pieces of music or voices(audio signals) from new digital video disks as well as from theconventional CDs. The recording systems for audio signals include acompression system and a linear PCM system. If one considered for avideo disk from which audio signals of pieces of music and voices can bereproduced by an exclusive audio player, it is effective to record databy the linear PCM technique as used for conventional CDs. It is verylikely that both low-class and high-class types of video disk playersbecome available on the market.

BRIEF SUMMARY OF THE INVENTION

Accordingly, it is an object of the present invention to provide a dataarranging method and a medium for data recording or transfer, which areeffective in recording or processing data or the like of a linear PCMsystem and which can record multi-channel signals of higher quality thanthat of the conventional CDs and can allow both low-class and high-classmachines to easily perform a reproduction process, and a signalprocessing apparatus for processing such data.

To achieve the above object, according to this invention, a system forrecording or transferring quantized data obtained by sampling onechannel or multi-channel signals in a time sequential manner andreproducing the quantized data handles a basic data structure in whichM-bit sample data of each channel signal is separated into a main wordconsisting of m1 bits on an MSB (Most Significant Bit) side and an extraword consisting of m2 bits on an LSB (Least Significant Bit) side, acollection of main words of 2n-th sample data of individual channels isarranged as a main sample S2n, a collection of main words of (2n+1)-thsample data of individual channels is arranged next as a main sampleS2n+1, a collection of extra words of 2n-th sample data of individualchannels is arranged as an extra sample e2n, and a collection of extrawords of (2n+1)-th sample data of individual channels is arranged as anextra sample e2n+1 (where n=0, 1, 2, . . . ).

With the above structure, a reproduction circuit is easily accomplishedin an inexpensive machine which reproduces only main words or only twochannels of main words while a reproduction circuit for extra words hasonly to be added to a main word reproduction circuit in a high-classmachine.

Additional objects and advantages of the invention will be set forth inthe description which follows, and in part will be obvious from thedescription, or may be learned by practice of the invention. The objectsand advantages of the invention may be realized and obtained by means ofthe instrumentalities and combinations particularly pointed out in theappended claims.

BRIEF DESCRIPTION OF THE DRAWINGS

The accompanying drawings, which are incorporated in and constitute apart of the specification, illustrate presently preferred embodiments ofthe invention, and together with the general description given above andthe detailed description of the preferred embodiments given below, serveto explain the principles of the invention.

FIGS. 1A-1D are explanatory diagrams showing a sample structure and thearrangement of samples for explaining a basic embodiment of thisinvention;

FIG. 2 is an explanatory diagram illustrating a relationship among thesamples in FIG. 1D, a frame, and a group;

FIGS. 3A and 3B are explanatory diagrams illustrating a relationshipbetween an audio frame and a sequence of packs according to thisinvention;

FIGS. 4A and 4B are diagrams showing general audio data arrangements ina 20-bit mode and a 24-bit mode;

FIG. 5 is an explanatory diagram illustrating the principle ofinterleaving;

FIGS. 6A and 6B are explanatory diagrams showing an example of thearrangement of packs and the structure of an audio pack in thisarrangement according to this invention;

FIG. 7 is an explanatory diagram depicting the detailed structure of anaudio pack;

FIG. 8 is an explanatory diagram exemplifying a list of sizes of linearPCM data in a packet, to which this invention is adapted;

FIG. 9 is an explanatory diagram illustrating procedures of generatingan audio pack;

FIG. 10 is a block structural diagram of a disk playing apparatus;

FIG. 11 is an explanatory diagram of a disk drive section;

FIG. 12 is an explanatory diagram of an optical disk;

FIG. 13 is an explanatory diagram illustrating the logical format of anoptical disk;

FIG. 14 is an explanatory diagram of a video manager in FIG. 13;

FIG. 15 is an explanatory diagram of a video object set in FIG. 14;

FIG. 16 is an explanatory diagram of a program chain;

FIG. 17 is a diagram showing one example of the basic circuit structureof an audio decoder according to this invention;

FIG. 18 is a diagram showing a second example of the basic circuitstructure of the audio decoder;

FIG. 19 is a diagram showing a third example of the basic circuitstructure of the audio decoder;

FIG. 20 is a diagram showing a fourth example of the basic circuitstructure of the audio decoder;

FIG. 21 is a table representing the contents of the pack header of theaudio pack;

FIG. 22 is a table illustrating the contents of the packet header of theaudio pack;

FIG. 23 is a block diagram showing mainly the audio data processingsystem incorporated in the disk playing apparatus;

FIGS. 24A-24D are diagrams showing a disk, a pit train, a sector train,and a physical sector, respectively;

FIGS. 25A and 25B are a diagram showing a physical sector and a tablerepresenting the contents of the physical sector, respectively;

FIGS. 26A and 26B are diagrams showing the structure of arecording/recorded sector; and

FIGS. 27A and 27B are diagrams illustrating an error-correction codeblock.

DETAILED DESCRIPTION OF THE INVENTION

Preferred embodiments of the present invention will now be describedwith reference to the accompanying drawings.

To begin with, a data arrangement by the linear PCM system in the datarecording system according to this invention will be discussed. Notethat 16 bits, 20 bits or 24 bits, for example, are arbitrarily used asquantization bits in linear PCM data. Further, audio modes includemonaural, stereo, 3 channel, 4 channel, 5 channel, 6 channel, 7 channeland 8 channel modes.

Suppose that there are eight channels (A to H) of audio signals. Thoseaudio signals are sampled at a sampling frequency of 48 KHz or 96 KHz tobe quantized. The following will describe an example where thequantization bits are 20 bits.

FIG. 1A shows how eight channels of audio signals A to H are sampled. Itis assumed that each sample is quantized to, for example, 20 bits. It isalso illustrated that each sample of 20 bits is separated into a mainword and an extra word.

The main words of the individual channels are indicated by largealphabet letters plus a suffix "n," and the extra words by smallalphabet letters plus the suffix "n", where n=0, 1, 2, 3, . . . ,indicates the sampling order. Each main word consists of 16 bits andeach extra word consists of 4 bits.

Individual samples are generated in the form of A0a0, A1a1, A2a2, A3a3,A4a4, and so forth for signal A; B0b0, B1b1, B2b2, B3b3, B4b4, and soforth for signal B; C0c0, C1c1, C2c2, C3c3, C4c4, and so forth for thesignal C; . . . ; H0h0, H1h1, H2h2, H3h3, H4h4, and so forth for signalH.

FIG. 1B illustrates the above word arrangement format as a sequence ofsamples in the case where those words are recorded on a recordingmedium.

First, each sample data consisting of 20 (=M) bits is separated to amain word of 16 (=m1) bits on the MSB side and an extra word of 4 (=m2)bits on the LSB side. Next, the zero-th (=2n-th) main words in theindividual channels are collectively arranged. Then, the first(=(2n+1)-th) main words in the individual channels are collectivelyarranged. Then, the zero-th (=2n-th) extra words in the individualchannels are collectively arranged. Then, the first (=(2n+1)-th) extrawords in the individual channels are collectively arranged. Note thatn=0, 1, 2, . . . .

A group of main words in the individual channels is one main sample.Likewise, a group of extra words in the individual channels is one extrasample.

With such a format employed, a data reproduction process by a low-costmachine (e.g., one which operates in a 16-bit mode) should handle onlymain words, while a data reproduction process by a high-cost machine(e.g., one which operates in a 20-bit mode) should handle both mainwords and their associated extra words.

FIG. 1C shows how individual samples are arranged by using the specificnumbers of bits for the main sample and extra sample.

In the form of such quantized linear PCM codes, the separation of a20-bit sample to a 16-bit main word and a 4-bit extra word can permitthe following. The machine which operates in the 16-bit mode can easilydiscard unnecessary portions by performing data processing in 8 bitunits in the areas of extra samples in the sample arrangement. This isbecause two extra samples are 4 bits×8 channels and 4 bits×8 channels,and those data can be processed (discarded) eight consecutive times in 8bit units.

The feature of this data arrangement is not limited to that of thisembodiment. In either case where there are an odd number of channels, orwhere an extra word consists of 8 bits, the total number of bits of twoconsecutive extra samples is an integer multiple of 8 bits, so that alow-cost machine which reproduces only main words can skip extra samplesby executing a discarding process n consecutive times 8 bits by 8 bitsin accordance with the mode.

Data in the as shown in FIG. 1B may then be subjected to a modulationprocess to be recorded on a recording medium. If data is to be recordedtogether with other control information and video information, it ispreferable that data should be recorded in the form that is easilymanaged on the time base in order to facilitate data handling andsynchronization. In this respect, the following frame formation,grouping and packet formation is useful.

FIG. 1D shows a sequence of audio frames. The unit of data over a givenreproduction time is 1/600 sec, which is one frame. In one frame, 80 or160 samples are assigned. With a sampling frequency of 48 KHz, onesample is 1/4800 sec and (1/48000)×80 samples=1/600 sec. With a samplingfrequency of 96 KHz, one sample is 1/9600 sec and (1/96000)×160samples=1/600 sec. Obviously, one frame consists of 80 samples or 160samples.

FIG. 2 shows a relationship between the aforementioned one frame and oneGOF (Group Of Frames). One frame consists of 80 or 160 samples and isdata of 1/600 sec, and one GOF consists of 20 frames. Thus, one GOF is(1/600) sec×20=1/30 sec, which is the frequency of one TV frame. Asequence of such GOFs is an audio stream. This unit, GOF, becomeseffective for synchronization with a video signal. As this frame isrecorded together with other control signals and video signals, it isdistributed to packets. The relationship between this packet and a framewill be described below.

FIG. 3A shows the relationship between the packet and frame.

DSI is data search information, V is a video object, A is an audioobject, and S is a sub picture object. Each block is called a pack. Onepack is defined as 2048 bytes. One pack includes a pack header, a packetheader, and a packet. Described in DSI is information for controllingeach a piece of data during playback, such as the start address and endaddress of each pack.

FIG. 3B shows only audio packs extracted. Although DSI packs, videopacks V, and audio packs A are actually mixed in the arrangement asshown in FIG. 3A, only audio packs A are illustrated in FIG. 3B to helpunderstand the relationship between a frame and packs. According to thestandards of this system, information is arranged so that it takes about0.5 sec to reproduce information between one DSI and the next DSI. Asone frame is 1/600 sec as mentioned above, 30 audio frames exist betweenone DSI and another DSI. The amount of data (D) of one frame variesdepending on the sampling frequency (fs), the number of channels (N),and the number of quantization bits (m).

When fs=48 KHz, D=80×N×m, and when fs=96 KHz, D=160×N×m.

Therefore, one frame should not necessarily correspond to one pack. Aplurality of frames or less than one frame may correspond to one pack.That is, the head of a frame may come in the middle of one pack as shownin FIG. 3B. Positional information of the head of a frame is describedin the pack header, and is described as the number of data counts(timings) from the pack header or DSI. When reproducing data from theaforementioned recording medium, the reproducing apparatus acquires aframe of audio packets, extracts data of a channel to be reproduced, andsupplies the data to the audio decoder to perform a decoding process.

FIG. 4A illustrates the relationship between a main word (16 bits) andan extra word (4 bits) in the 20-bit mode, generally showing theaforementioned data arrangement, and FIG. 4B illustrates therelationship between a main word (16 bits) and an extra word (8 bits) inthe 24-bit mode.

As shown in FIGS. 4A and 4B, sample data has the aforementioned framestructure and pack structure with an integer multiple of twin pairs ofsamples. Each pair includes a main sample and an extra sample.

The foregoing description has been given on the premise that nointerleave process is performed in the signal format. When there is ascratch on the recording medium or consecutive drops of data during datatransfer, interleaving can reduce the consecutive signal losses, if ithas been performed. It is known that interleaving permits approximateinterpolation of lost sample data.

FIG. 5 illustrates the principle of interleaving and deinterleaving forthe above-described format. According to the data arrangement of thisinvention, even when interleaving is executed, a low-cost machine caneasily deinterleave only main words. This feature allows the circuit tobe simplified.

This example employs a delay interleave technique with an interleavelength D of 2k samples. In the figure, S means one main sample, and themain samples are S0=A0, B0, . . . H0; S1=A1, B1, . . . H1; S2=A2, B2, .. . H2; and Sj=Aj, Bj, Cj, . . . Hj. The letter "e" means an extrasample, and the extra samples are e0=a0, b0 . . . h0; e1=a1, b1, . . .h1; e2=a2, b2, . . . h2; and ej=aj, bj, cj, . . . hj. Even main samplesare input to a delayless transmission system L11, and odd main samplesare input to a delay transmission system L12. Even extra samples areinput to a delayless transmission system L13, and odd extra samples areinput to a delay transmission system L14.

The delay amount of extra samples which each consists of 4 bits, can beone fourth the delay amount of main samples (16 bits), and the delayamount of extra samples each consisting of 8 bits, can be half of thedelay amount of main samples (16 bits). Therefore, the delaytransmission system L14 is designed to be able to switch the delayamount between the 20-bit mode and the 24-bit mode.

Columns of the individual samples on the input side of the transmissionsystems in FIG. 5 maintain the format which has been discussed withreference to FIG. 1B. With the columns of samples synchronized, theindividual samples are input to the associated transmission systems. Asa result, a two-dimensional arrangement of samples, as seen on theright-hand side of the individual transmission systems, is acquired.Although the data contents of columns in the two-dimensional array aredifferent from those before interleaving, this array still containscombinations of two main samples and two extra samples in the verticaldirection.

In executing the deinterleave process, even columns of main samples areinput to a delay transmission path while odd columns of main samples areinput to a delayless transmission path. Likewise, even columns of extrasamples are input to a delay transmission path while odd columns ofextra samples are input to a delayless transmission path. Thisprocessing can provide the original sample arrangement. In the 16-bitmode, only the transmission systems for main samples should be used.

On the reproduction side, a machine which reproduces only main samplesshould have a deinterleave circuit which handles only main samples. Toreproduce only a specific channel, a deinterleave circuit which handleswords in sample data of that specific channel is used.

As described above, this invention can provide a data arranging methodand a medium for recording or transferring multi-channel data of thelinear PCM system which can allow both low costs and high costs machinesto perform a reproduction process, and a processing apparatus whichprocesses such data.

FIG. 6A illustrates the arrangement of packs. Each pack includes apacket.

DSI is data search information, V is a video object, A is an audioobject, and S is a sub picture object. Each block is called a pack. Thesize of one pack is set at 2048 bytes. One pack includes one packet, andconsists of a pack header, a packet header and a packet data section.Described in DSI is information for controlling each piece of dataduring playback such as the start address and end address of each pack.

FIG. 6B shows only audio packs A extracted. Although DSI packs, videopacks, and audio packs are actually mixed in the arrangement, as shownin FIG. 6A, only audio packs are illustrated in FIG. 6B to helpunderstand packs. The standards of this system define that the amount ofinformation arranged between DSIs should be equivalent to about 0.5 secwhen information between DSIs is reproduced. As mentioned above, onepack consists of a pack header, a packet header, and a packet datasection.

Described in the pack header and the packet header as informationnecessary to reproduce audio data, such as the size of an audio pack,presentation time stamp for synchronization with the reproduction outputof video data, an identification (ID) code of a channel (stream),quantization bits, a sampling frequency, and start address and endaddress of data.

Next, audio data inserted in this packet has twin pairs of samples. Eachpair has two main samples, as and two extra samples shown in FIGS.1A-1C.

FIG. 7 shows an enlarged audio pack. Arranged in the data section ofthis audio pack are twin pairs of samples with the top twin pair ofsamples (A0-H0, A1-H1) located at the top of the data area. The numberof bytes in one pack is set at 2048 bytes. As samples are variablelength data, 2048 bytes should not necessarily be equal to an integermultiple of the byte length of twin pairs of samples. Therefore, theremay be a case where the maximum byte length of one pack differs from thebyte length of a twin pair of samples×integer number. In this case, thebyte length of a pack is greater than or equal to a twin pair ofsamples×integer number. If a part of a pack remains, a stuffing byte isinserted in the pack header, when the remainder is equal to or less than7 bytes, or padding packet is inserted at the end of the pack, when theremainder exceeds 7 bytes.

Audio information in this pack format can easily be handled at the timeof reproduction.

Since the top audio data in each pack is the top twin pair of samples ormain samples, the reproduction process becomes easier when reproductionis executed with the proper timing. This is because the reproductionapparatus acquires data and performs data processing pack by pack. Ifsamples of audio data are located over two packs, the two packs shouldbe acquired and, the audio data should be decoded after integration.This complicates the processing. When the top audio data in each pack isalways the top twin pair of samples and audio data is grouped pack bypack as in this invention, timing should be taken only for one pack,thus facilitating the data processing. Further, the packet-by-packetdata processing simplifies the authoring system (aiding system), whichcan simplify software for processing data.

At the time of special reproduction or the like, video data may besubjected to thinning or interpolation. In such a case, since audio datais permitted to be handled packet by packet, it is possible torelatively easily control the reproduction timing. Further, software forthe decoders need not be complicated.

Although samples are generated with each sample separated into the upper16 bits and the lower 4 bits in the above-described system, data shouldnot necessarily take such a format as long as linear PCM audio data issampled.

With the data length of an extra sample set to 0, for example, a trainof data becomes a sequence of main samples which is the general dataformat. In this case, no extra samples are present, so there is no needto generate twin pairs of samples and main samples alone are formed intopackets.

FIG. 8 shows a list of the sizes of linear PCM data when linear PCM dataare arranged in a packet in units of twin pairs of samples as discussedabove. The data sizes are shown as the number of maximum samples to befit in one pack, separately for the monaural (mono), stereo, andmulti-channel modes. Each group shows the data sizes for the respectivenumbers of quantization bits. Because twin pairs of samples are taken asunits, every number of samples in one packet is an even number. As thenumber of channels increases, the number of bytes increases accordingly,so that the number of samples in one packet decreases. When the numberof quantization bits is 16 bits and the mode is the monaural mode, thenumber of samples in one packet is 1004, and number of bytes is 2008with the stuffing byte of 5 bytes, which indicates that there are nopadding bytes. Note however, that the first packet has the stuffingbytes of 2 bytes. This is because 3-byte attribute information may beaffixed to the header of the first packet.

With the number of quantization bits being 24 bits and in the stereomode, stuffing 6 bytes is given to the top packet and padding 9 bytes isgiven to the subsequent packets.

FIG. 9 illustrates the operational procedures of the apparatus whichgenerates packs.

Suppose that audio signals of each channel are samples to produce thesamples as shown in FIG. 1B, which are stored in the memory. In stepS11, samples are acquired one by one. In step S12, it is determined ifthe number of bytes has reached the capacity of a packet (2010 bytes).When 2010 bytes are reached, those samples up to that sample are packed(step S13).

When the number of bytes has not reached the capacity of a packet (2010bytes), the flow proceeds to step S14 where it is determined if thenumber of bytes of the acquired samples exceeds 2010 bytes. When it doesnot exceed 2010 bytes, the flow returns to step S11. When it exceeds2010 bytes, on the other hand, the last acquired sample is returned tothe position of step S11 and the difference between the number ofremaining bytes and 2010 bytes is computed in step S15. It is thendetermined if this difference R exceeds 8 bytes (step S16). When thedifference R exceeds 8 bytes, padding is performed (step S17) toconstruct a packet, whereas when the difference R is equal to less than7 bytes, stuffing is performed (step S18) to construct a packet.

The reproduction apparatus which reproduces the above-discussed datawill be briefly described.

FIG. 10 shows an optical disk player. FIG. 11 shows the basic structureof a disk drive section 501 which drives an optical disk 10 on which theabove-described audio stream is recorded. FIG. 12 presents a diagram forexplaining an example of the structure of the optical disk 10.

The optical disk player in FIG. 10 will now be discussed.

The optical disk player has a key operation/ display section 500. Theoptical disk player is connected to a monitor 11 and speakers 12. Datapicked up from the optical disk 10 is sent via the disk drive section501 to a system processing section 504. The picked-up data from theoptical disk 10 includes picture data, sub picture data, and audio data,for example, which are separated in the system processing section 504.The separated picture data is supplied via a video buffer 506 to a videodecoder 508, the sub picture data is supplied via a sub picture buffer507 to a sub picture decoder 509, and the audio data is supplied via anaudio buffer 507 to an audio decoder 513. The picture signal decoded bythe video decoder 508 and the sub picture signal decoded by the subpicture decoder 509 are combined by a synthesizing section 510, and theresultant signal is converted to an analog picture signal by a D/Aconverter 511. This analog picture signal is then sent to the monitor11. The audio signal decoded by the audio decoder 513 is converted by aD/A converter 514 to an analog audio signal which is in turn supplied tothe speakers 12.

The entire player is controlled by a system CPU 502. That is, the systemCPU 502 can exchange control signals, timing signals, and the like, withthe disk drive section 501, the system processing section 504, and thekey operation/display section 500. Connected to the system CPU 502 is asystem ROM/RAM 503 in which fixed programs for allowing the system CPU502 to execute data processing are stored. Management data or the like,which is reproduced from the optical disk 10, can also be stored in thesystem ROM/RAM 503.

A data RAM 505, connected to the system processing section 504, is usedas a buffer when the aforementioned data separation, error correction,or the like, is executed.

The disk drive section 501 in FIG. 11 will now be discussed.

A disk motor driver 531 drives a spindle motor 532. As the spindle motor532 rotates, the optical disk 10 turns and data recorded on the opticaldisk 10 can be picked up by an optical head section 533. The signalpicked up by the optical head section 533 is sent to a head amplifier534 whose output is input to the system processing section 504.

A feed motor 535 is driven by a feed motor driver 536. The feed motor535 drives the optical head section 533 in the radial direction of theoptical disk 10. The optical head section 533 is provided with a focusmechanism and a tracking mechanism to which drive signals from a focuscircuit 537 and a tracking circuit 538 are, respectively, supplied.

Control signals are input to the disk motor driver 531, the feed motordriver 536, the focus circuit 537, and the tracking circuit 538 from aservo processor 539. Accordingly, the disk motor 532 controls therotation of the optical disk 10 in such a way that the frequency of thepicked-up signal becomes a predetermined frequency, the focus circuit537 controls the focus mechanism of the optical system in such a waythat the optical beam from the optical head section 533 forms theoptimal focal point on the optical disk 10, and the tracking circuit 538controls the tracking mechanism in such a way that the optical beam hitsthe center of the desired recording track.

The structure of the optical disk 10 shown in FIG. 12 will now beexplained.

The optical disk 10 has information recording areas 22 around clampareas 21 on both sides. The information recording area 22 has a lead-outarea 23 where no information is recorded at the outer periphery, and alead-in area 24 where no information is recorded at the boundary withthe associated clamp area 21. Between the lead-out area 23 and thelead-in area 24 lies a data recording area 25.

Tracks are continuously formed in the data recording area 25 in a spiralform. The tracks are separated into a plurality of physical sectorswhich are given serial numbers. Signal spots on tracks are formed aspits. For a read-only optical disk, a sequence of pits is formed on atransparent substrate by a stamper, and a reflection film is formed onthe pitted surface to form a recording layer. A double-disk type opticaldisk has two disks adhered together via an adhesive layer, yielding acomposite disk, in such a manner that the recording layers face eachother.

The logical format of the optical disk 10 will now be discussed.

FIG. 13 shows the logical format of the information sections of theinformation recording area 25. This logical format is determined inconformity to specific standards, such as micro UDF and ISO 9660. In thefollowing description a logical address means a logical sector number(LSN) which is determined by the micro UDF and ISO 9660, and logicalsectors are the same size as the aforementioned physical sectors. Eachlogical sector has 2048 bytes. It is assumed that serial logical sectornumbers (LSN) are given to the logical sectors in the ascending order ofthe physical sector numbers.

The logical format is a hierarchical structure and has a volume and filestructure area 70, a video manager 71, at least one video title set 72,and an other recording area 73. Those areas are differentiated at theboundaries of the logical sectors. As mentioned above, the size of onelogical sector is 2048 bytes. The size of one logic block is also 2048bytes, so that one logical sector is defined as one logic block.

The file structure area 70 is equivalent to a management area which isdefined by the micro UDF and ISO 9660, and data in the video manager 71is stored in the system ROM/RAM section 52 via the description in thisarea 70. Information for managing the video title sets is described inthe video manager 71, which consists of a plurality of files 74 startingwith file #0. Recorded in each video title set 72 are compressed videodata, sub picture data, audio data, and playback control information forreproducing those data. Each video title set 72 consists of a pluralityof files 74, which are also differentiated at the boundaries of thelogical sectors.

Recorded in the other recording area 73 is information which is usedwhen the information in the video title set is used or information whichis exclusively used.

The video manager 71 will be described below with reference to FIG. 14.

The video manager 71 consists of video manager information (VMGI) 75, avideo object set for a video manager information menu (VMGM₋₋ VOBS) 76and a backup of video manager information (VMGI₋₋ BUP) 77.

Stored in the VMGM₋₋ VOBS 76 are video data, audio data and sub picturedata for the menu which is associated with the volume of the opticaldisk. The VMGM₋₋ VOBS 76 can provide descriptive information, given byvoices and a sub picture in association with each of titles in thevolume, and the selection display for the titles. When Englishconversations for learning English are recorded on the optical disk, forexample, the title name of each English conversation and examples of alesson are reproduced and displayed while a theme song is acousticallyreproduced, and each sub picture shows which text of which level or thelike. The lesson numbers (levels) are displayed as selection items whichshould be selected by a listener. The VMGM₋₋ VOBS 76 is used for such ausage.

FIG. 15 illustrates a video object set (VOBS) 82.

There are two types of video object sets for a menu and one type ofvideo object set for video titles. The three types of video abject sethave similar structures.

The VOBS 82 is defined as a set of one or more video objects (VOB's) 83,which are used for the same purpose. Normally, the VOBS for a menuconsists of video objects (VOB's) for displaying a plurality of menuscreens, while the VOBS for a video title set consists of VOB's fordisplaying normal moving pictures or the like.

Each VOB is given an ID number (VOB₋₋ IDN#j), which is used to identifythat VOB. One VOB consists of one cell or a plurality of cells 84.Likewise, each cell is given an ID number (C₋₋ IDN#j), which is used toidentify that cell. The video object for a menu may be comprised of asingle cell.

Further, one cell consists of one or a plurality of video object units(VOBU's). A single VOBU is defined as a sequence of packs having anavigation pack 86 (nav pack) at the top. One VOBU is defined as a setof all packs recorded between the nav pack 86 (including theaforementioned DSI) and the next nav pack 86.

The playback time for the VOBU is equivalent to the playback time forvideo data which consists of a single GOP (Group Of Picture) or aplurality of GOP's included in this VOBU, and is defined to be equal toor greater than approximately 0.4 sec and equal to or less than 1 sec.The MPEG standards define one GOP as compressed image data equivalent tothe playback time of about 0.5 sec. According to the MPEG standards,therefore, about 0.5 sec of audio information and picture informationcan be arranged.

One VOBU has the aforementioned nav pack 86 at the top, followed byvideo packs (V packs) 88, sub picture packs (SP packs) 90 and audiopacks (A packs) 91 arranged in a certain order. A plurality of V packs88 in one VOBU 85 has compressed image data whose playback time is equalto or less than 1 sec, in the form of one GOP or a plurality of GOP's.Audio signals corresponding to this playback time are compressed andarranged as A packs 91. The sub picture data used within this playbacktime is compressed and is arranged as SP packs 90. It is to be notedthat audio signals are recorded with, for example, eight streams of dataas a pack, and sub pictures are recorded with, for example, thirty twostreams of data as a pack.

One stream of audio signals is data encoded by one kind of codingsystem, and consists of eight channels of linear PCM quantized data of20 bits, for example.

Returning to FIG. 14, the VMGI 75 describes information for searchingfor a video title, and includes at least three tables 78, 79 and 80.

A video manager information management table (VMGI₋₋ MAT) 78 describesthe size of the VMG 71, the start address of each information in thevideo manager, attribute information associated with the video objectset for a video manager menu (VMGM₋₋ VOBS), and the like.

A title search pointer table (TT₋₋ SRPT) 79 describes entry programchains (EPGC) of the video titles included in the volume of the opticaldisk which are selectable in accordance with the title number inputthrough the key operation/display section of the apparatus.

The program chains will now be discussed referring to FIG. 16. Eachprogram chain 87 is a set of program numbers for reproducing the storyof a certain title. A chapter of the story of one title or the storyitself is completed as program chains are continuously reproduced. Oneprogram number consists of a plurality of cell ID numbers each of whichcan specify a cell in the VOBS.

A video title set attribute table (VTS₋₋ ART) 80 describes attributeinformation which is determined by video title sets (VTS) in the volumeof the optical disk. The attribute information includes the number ofVTS, the number, the video compression system, the audio coding mode,and the display type of sub pictures.

According to the packet system according to this invention, as describedabove, audio data at the top of each packet is always at the top ofsample data, and packets can be treated as units, so that the timingprocessing for processing audio data and a sequence of processes of thistiming processing becomes easier.

A description will now be given of the audio decoder which reproducesdata that is arranged and recorded in the above-described form.

FIG. 17 shows the basic structure of the audio decoder 513.

The illustrated decoder can reproduce data in all the modes for thenumbers of channels and the numbers of bits of samples, as shown in FIG.8. Input data is such that the number of quantization bits of every oneof eight channels is 24 bits.

A sequence of samples as discussed with reference to FIG. 1 iscontinuously input to an input terminal 710. This sequence of samples isgiven to the input terminal, 711, of a switch SW0. The switch SW0 hasdistribution terminals for the individual samples of channels An to Hnand an to hn. The terminals which are associated with samples of theindividual channels are given the same reference numerals asrepresentative samples. The representative samples are samples A0 to H0,A1 to H1, a0 to h0, and a1 to h1.

It is assumed that the terminals A0 to H0 and A1 to H1 are 16-bitterminals, and the terminals a0 to h0 and a1 to h1 are 4-bit terminals.The extra sample may consist a total of eight bits so that two sets of4-bit terminals, a0 to h0 and a1 to h1, are prepared. The 16-bitterminal A0 is connected to the upper bits (16 bits) of a memory MA0,and the associated 4-bit terminals a0 and a0 are connected to the lowerbits (8 bits) of the memory MA0 via respective switches j1 and j2. The16-bit terminal B0 is connected via a switch JB to the upper bits of amemory MB0, and the associated 4-bit terminals b0 and b0 are connectedto the lower bits of the memory MB0 via respective switches j1 and j2.The 16-bit terminal C0 is connected via a switch JC to the upper bits ofa memory MC0, and the associated 4-bit terminals c0 and c0 are connectedto the lower bits of the memory MC0 via respective switches j1 and j2.Likewise, the other terminals D0 to H0, D1 to H1, d0 to h0, and d1 to h1are connected to associated memories MD0 to MH1.

As a result, the individual channels are distributed to the memories MA0to MH1. The output terminals of the memories MA0 and MA1 are connectedto terminals TA0, Ta0, Ta0, TA1, Ta1 and Ta1 of an A channel outputswitch SWA. TA0 and TA1 are 16-bit terminals, and Ta0, Ta0, Ta1 and Ta1are 4-bit terminals. Likewise, the output terminals of the memories MB0and MB1 are connected to terminals TB0, Tb0, Tb0, TB1, Tb1 and Tb1 of aB channel output switch SWB. TB0 and TB1 are 16-bit terminals, and Tb0,Tb0, Tb1 and Tb1 are 4-bit terminals. The output terminals of the othermemories are likewise connected to the associated output switches.

The operation of the audio decoder 513 will now be discussed.

Samples S0, S1, e0, e1, . . . , which are arranged forrecording/transfer and are to be input to the switch SW0, can beexpressed as A0, B0, . . . , H0, A1, B1, . . . , H1, a0, b0, . . . , h0,a1, b1, . . . , h0 as samples of the individual channels. Each of themain words of each channel consists of 16 bits, and each extra wordconsists of 8 bits. Suppose that the switches of the circuit are allclosed. As the rotary switch SW0 is sequentially switched from thetopmost contact, associated samples are transferred to the memories MA0to MH1. In this manner, twin pairs of samples are cyclically stored inthe memories MA0 to MH1 by the action of the rotary switch SW0.Thereafter, samples of the desired channel among those samples stored inthe memories MA0 to MH1 are read via the associated rotary switch. Themain sample and the extra sample in each read sample are decoded andthen combined for the subsequent processing.

Let us pay attention to the reading of the channel A. With the rotaryswitch SWA at the topmost 16-bit contact position, the 16-bit sample A0is read. Then, samples a0 having a total of 8 bits are read at two 4-bitcontact positions. At the next 16-bit contact position, the 16-bitsample A1 is read. Then, samples a1 having a total of 8 bits are read attwo 4-bit contact positions. As the rotary switch SWA rotates once, twinpairs of samples A0, a0 and A1, a1 of the channel A are read out. Inthis manner, twin pairs of samples of the channel A are obtained in atime sequential form. Thereafter, with regard to the other channels B, Cand so forth, samples are likewise read. Because twin pairs of samplesare processed as each of the rotary switches SW0, SWA, . . . , and SWHmakes one turn, the rotational period should be a half of the samplingfrequency (fs/2).

FIG. 18 illustrates another embodiment of the audio decoder.

The illustrated embodiment processes data in the case where there aretwo channels and the number of quantization bits of each sample is 20bits. This circuit differs from the one shown in FIG. 17 in the statusesof the switches JB-JH, j1 and j2. Therefore, same reference numerals aregiven to those components which are the same as the correspondingcomponents of the circuit in FIG. 17.

Samples S0, S1, e0, e1, and so forth are expressed as A0, B0, A1, B1,a0, b0, a1, b1, and so forth, as a sequence of samples of the individualchannels. Each main sample of each channel consists of 16 bits, and eachextra sample consists of 8 bits. As illustrated, only the switch JB isclosed, and the switches JC to JH are open. With regard to thoseswitches j1 and j2 which are associated with the extra samples a0, b0,a1 and b1, as illustrated, only the switches j1 are closed and the otherswitches are open. Those switches j1 and j2 which are associated withthe other extra samples c0, . . . , h0, c1, . . . , h1 are all open.

When the rotary switch SW0 distributes input data in synchronism withthe data input, data to be transferred are A0, B0, A1, B1, a0 (4 bits),b0 (4 bits), a1 (4 bits) and b1 (4 bits). The action of the rotaryswitch SW0 allows the samples to be input to only the memories MA0, MB0,MA1 and MB1 in the illustrated order.

On the output side, outputs are obtained from those of the memories MA0to MH1 which are associated with the channels A and B are acquired. Data0 is output from the memories associated with the other channels. Of theswitches j1 and j2 on the reading side, the switches j1 are closed andthe switches j2 are open. Accordingly, a 4-bit extra sample is read outfollowing a 16-bit main sample. As regards the channel A, as the switchSWA is switched, data of the channel A is sequentially output in theorder of A0, a0 (4 bits), A1 and a1 (4 bits).

The settings of the individual switches and the switching operations inthe above-described embodiment are programmably set in accordance withthe number of channels of audio streams and the number of quantizationbits of each sample. Such a signal processing mode is described in thevideo title set attribute table shown in FIG. 14 and the packet headershown in FIG. 7. In other words, audio data included in an audio packetbeing linear PCM data, the audio frame number, the number ofquantization bits, the sampling frequency, the audio channel number,etc. are described.

The decoders illustrated in FIGS. 17 and 18 can handle all the modes andare so-called full decoders that are adaptable in a high-cost machinewhich can reproduce all the channels.

The concept of this invention relates to a data arranging method, arecording/reproducing method and a processing apparatus, which canhandle various kinds of modes established by multifarious combinationsof the number of channels and the number of quantization bits. The dataarrangement can be adapted to the aforementioned high-cost machine aswell as an inexpensive machine which meets the demand for a lower cost,e.g., one which reproduces only 16-bit data of two channels in everymode. Such a machine advantageously has a smaller circuit scale than thehigh-cost machine.

Although the switches which are used to distribute individual samplesand acquire samples from the associated memories are illustrated asmechanical switches, they all are of electronic circuits.

An audio decoder in a low cost player will now be described. This audiodecoder processes 16-bit data of only the channels A and B. Inputsamples are of eight channels and the number of quantization bits is 24bits.

A sequence of samples as discussed with reference to FIG. 1 iscontinuously input to an input terminal 810 in FIG. 19. This sequence ofsamples is given to the input terminal, 811, of a switch SW0. The switchSW0 has distribution terminals for the individual samples of channels Anto Hn and an to hn. The terminals which are associated with samples ofthe individual channels are given the same reference numerals asrepresentative samples, which are samples A0 to H0, A1 to H1, a0 to h0and a1 to h1.

It is assumed that the terminals A0 to H0 and A1 to H1 are 16-bitterminals, and the terminals a0 to h0 and a1 to h1 are 4-bit terminals.Since the extra sample may consist a total of eight bits, two sets of4-bit terminals, a0 to h0 and a1 to h1, are prepared.

In this decoder, however, only the terminals A0 and A1, and B0 and B1are respectively connected to the memories MA and MB, with the otherterminals C0-H0 and c0-h0 grounded. The switch SW0 may be designed inthis manner, or may be designed to have only those systems associatedwith the channels A and B from the beginning.

The switches SWA and SWB are for reading data from the memories MA andMB in the units of 16 bits. Those switches SWA and SWB operate in such away that output data are matched with one another.

The operation of this audio decoder will now be discussed.

Samples S0, S1, e0, e1, . . . , which are arranged for therecording/transfer purpose and are to be input to the switch SW0 can beexpressed as A0, B0, . . . , H0, A1, B1, . . . , H1, a0, b0, . . . , h0,a1, b1, . . . , h0 as samples of the individual channels. Each mainsample of each channel consists of 16 bits, and each extra word consistsof 8 bits. The switches of the circuit are all closed. As the rotaryswitch SW0 is sequentially switched from the topmost contact, associatedsamples are transferred to the memories MA0 and MB1. The other samplesare all discarded.

Thereafter, the samples stored in the memories MA0 and MB1 are readthose of the channels A and B.

Because two samples are processed as the rotary switch SW0 turns once,the rotational period should be a half of the sampling frequency fs.Because one sample is read as each of the rotary switches SWA and SWBturns once, the frequency is fs.

Another audio decoder in a low cost player will now be discussed. Thisaudio decoder processes 16-bit data of only the channels A and B. Inputsamples are of two channels and the number of quantization bits is 20bits.

A sequence of samples as discussed with reference to FIG. 1 iscontinuously input to the input terminal 810 in FIG. 20. This sequenceof samples is given to the input terminal 811 of the switch SW0. Theswitch SW0 has distribution terminals for the individual samples ofchannels An to Hn and an to hn. The terminals which are associated withsamples of the individual channels are given the same reference numeralsas representative samples, which are samples A0, B0, A1, B1, a0, b0, a1and b1.

The terminals A0, B0, A1 and B1 are 16-bit terminals, and the terminalsa0, b0, a1 and b1 are 4-bit terminals. To cope with the modes for twochannels and the quantization bits of 20 bits, only the switch JB may isclosed and the switches JC-JH are open. Those switches j1 and j2 whichare associated with the terminals a0, b0, a1 and b1 are closed andswitches j3-j16 associated with the other terminals are open.

As the rotary switch SW0 is sequentially switched in the abovesituation, no data transfer is performed. And only the main samples A0,B0, A1 and B1 are transferred to the memories MA and MB. Regarding theextra samples a0, b0, a1 and b1, since their associated switches aregrounded, those extra samples are discarded. The operation of readingsamples from the memories MA and MB is carried out in the same manner asdone in the previously described embodiment.

Although the foregoing description of the low cost machine has beengiven with reference to two modes, data of two channels can be acquiredin every mode in accordance with the selective open or closed states ofthe switches. The particular point that should be noted is thatprocessing for extra samples is executed 8 bits by 8 bits. Theabove-described data arrangement makes the number of bits of one pair ofextra samples an integer multiple of 8 bits regardless of the number ofchannels, even if each extra word of each channel consists of 4 bits.Even when extra samples are to be discarded in a low cost decoder,therefore, 8-bit processing is possible.

Since the main words of extra samples each consist of 16 bits, they canall be processed 8 bits by 8 bits, which is very advantageous indesigning a specific circuit.

Each audio pack has a pack header. As shown in FIG. 21, the pack headerconsists of a pack start code (4 bytes), a system clock reference (SCR)(6 bytes), a program multiplexing rate (3 bytes) and a pack stuffinglength (1 byte). The SCR represents the time required to fetch thisaudio pack. If the value of the SCR represents is shorter than areference value in the disk playing apparatus, the audio pack will bestored into the audio buffer. The control circuit refers to the packstuffing length and determines an read address on the basis of the packstuffing length.

FIG. 22 shows the contents of the packet header of an audio packet. Thepacket header includes a packet₋₋ start₋₋ code prefix indicative of thestart of a packet, a stream ID indicating what data the packet has, anddata indicative of the length of the packet stream. Also described inthe packet header are various kinds of information of packet elementarystream (PES), such as a flag indicating the inhibition or permission ofcopy, a flag indicating if the information is original one or copiedone, and the length of the packet header. A presentation time stamp(PTS) for synchronization of the output timing of this packet with thatof other video data or sub picture is further described in the packetheader. Further, information, such as a flag indicating if there is anydescription on a buffer and the buffer size, is described in the firstpacket in the first field in each video object.

The packet header also has stuffing bytes of 0 to 7 bytes. The packetheader further has a sub stream ID indicating an audio stream, linearPCM or other compressing type, and the number of audio stream. Furtherdescribed in the packet header are the number of frames of audio datawhose first byte is located in this packet. Furthermore, a pointer for aunit to be accessed first is described by the number of logic blocksfrom the last byte of this information. Thus, the pointer indicates thefirst audio frame to be decoded first at the time described by the PTS.The pointer indicates the first byte address of that audio frame.Further described in the packet header are an audio emphasis flagindicating whether or not to be emphasized high frequency band, a muteflag for providing mute when audio frame data are all 0, and a framenumber indicative of the frame in an audio frame group (GOF) whichshould be accessed first. Control information, such as the length of aquantized word or the number of quantization bits, the samplingfrequency, the number of channels and the dynamic range, is alsodescribed.

The above header information is analyzed by a decoder control section(not shown) in the audio decoder. The decoder control section switchesthe signal processing circuit in the decoder to the signal processingmode which is associated with currently acquired audio data. Theswitched modes are as discussed with reference to FIGS. 17 to 20.Information like this header information is also described in the videomanager, so that when such information is read at the initial stage ofthe reproducing operation, the information need not be read againthereafter for the reproduction of the same sub stream. The reason whymode information necessary to reproduce audio data is described in theheader of each packet as mentioned above is because a receiving terminalcan identify the mode of the audio data whenever reception starts in thecase a sequence of packets is transferred by a communication path.

FIG. 23 is a block diagram of the audio data processing systemincorporated in the disk playing apparatus, illustrating the systemprocessing section 504 and the audio decoder 513 in more detail thanFIG. 10.

In the system processing section 504, an input high-frequency signal(read signal) is supplied to a sync detector 601. The detector 601detects and extracts a sync signal from the read signal and generates atiming signal. The read signal now containing no sync signal is input toa 8-16 demodulator 602, which demodulates the 16-bit signal into a trainof 8-bit data. The 8-bit data is input to an error correcting circuit603. The data output from the circuit 603, which is free of errors, isinput to a demultiplexer 604. The demultiplexer 604 processes the data,recognizing the video pack, the sub-picture pack, and the audio packaccording to the reference of the stream ID. These packs are suppliedfrom the demultiplexer 604 to the video decoder 508, the sub-picturedecoder 509 and the audio decoder 513.

Meanwhile, the audio pack is fetched into an audio buffer 611, and thepack header and packet header of the audio pack are fetched into acontrol circuit 612. The control circuit 612 recognizes the contents ofthe audio pack, i.e., the start code, stuffing length, packet start codeand stream ID of the audio pack. Further, the control circuit 612recognizes the sub-stream ID, the first access point, number ofquantized audio bits, number of channels and sampling frequency. Thestuffing byte length and the padding packet length are determined fromthese data items thus recognized, on the basis of the table shown inFIG. 8.

The control circuit 612 recognizes the packet of linear PCM based on thesub-stream ID.

As a result, the control circuit 612 can identify the extraction addressof the audio data stored in the audio buffer 611. When controlled by thecircuit 612, the audio buffer 611 outputs samples such as samples S0,S1, e0, e1, S2, S3, . . . The control circuit 612 can recognize thenumber of stuffing bytes and/or the number of padding packets after ithas recognized at least the number of quantized bits, the samplingfrequency, and the number of audio channels. The circuit 612 can extractdata based on these recognized data items.

The samples output from the audio buffer 611 are supplied to a channelprocessor 613. The processor 613 has a structure of the type shown inFIGS. 17 to 20. Its operating mode is controlled by the control circuit612.

The audio packet, the video packet, the sub-picture packet and therecording tracks of the optical disk, all described above, have aspecific physical relationship, which will be explained below.

When a part of the recording surface of an optical disk 10 shown in FIG.24A is magnified, trains of pits are seen as illustrated in FIG. 24B. Aset of pit trains constitute a sector as seen in FIGS. 24C and 24D,which are two other magnified views of the optical disk 10. The sectorsare sequentially read by the optical head. Then the audio packets arereproduced in real time.

The sectors will be described with reference to FIGS. 25A and 25B. Asshown in FIG. 25B, a sector in which audio data is recorded, consists of13×2 frames. One sync code is assigned to each sector. Although theframes are shown in FIG. 25B as if sequentially arranged in rows andcolumns, they are sequentially arranged in a single row on one track.More specifically, the frames having sync codes SY0, SY5, SY1, SY5, SY2,SY5, . . . are arranged in the order they are mentioned.

The sync code assigned to one frame consists of 32 bits (16 bits×2), andthe data recorded in one frame consists of 1456 bits (16 bits×91). Thismeans that the sector is expressed by 16-bit modulated code, since16-bit data items obtained by modulating 8-bit data items are recordedon the optical disk. Also recorded in each sector is a modulatederror-correction code.

FIG. 26A shows a sector in which there are 8-bit data items obtained bydemodulating the 16-bit data items recorded in the physical sectordescribed above. The amount of data in this sector is: (172+19)bytes×(12+1) lines. Each line contains a 10-byte error-correction code.One correction code is provided for each line. When twelve correctioncodes for twelve lines, respectively, are collected, they function as anerror-correction code for twelve columns.

The data recorded in one recording/recorded sector becomes a data blockof the type shown in FIG. 26B when the error-correction signal isremoved from it. The data block consists of 2048-byte main data, 6-bytesector ID, a 2-byte ID error-detection code, 6-byte copyright managementdata, and a 4-byte error-detection code. As FIG. 26B shows, the sectorID, ID error-detection code and the copyright management data are addedto the head of the main data, whereas the error-detection code is addedto the end of the main data. The 2048-byte main data is one pack asdefined above. A pack header, packet header, and audio data aredescribed in the pack, in the order mentioned from the head of the pack.In the pack header and the packet header there are described variousitems of guide information which will be used to process the audio data.

As indicated above, one packet which consists of audio samples arrangedin a specific way is recorded in each recording/recorded sector on thedisk. The audio decoder can reproduce linear PCM data in a desiredmanner despite that the PCM data is recorded in one recording/recordedsector. This is because the start part of the audio data contained inany pack is the start part of the main sample, and also because the packheader and the packet header contain control data sufficient for theaudio decoder to process audio data.

An error-correction code (ECC) block will be described, with referenceto FIGS. 27A and 27B.

As shown in FIG. 27A, the ECC block consists of 16 recording/recordedsectors. As shown in FIG. 26A, each sector can record 12 lines of data,each line being a 127-byte data item. A 16-byte outer parity (PO) isadded to each column, and a 10-byte inner parity (PI) is added to eachline. As shown in FIG. 27B, the 16-byte outer parity (PO) isdistributed, one bit to each line. As a result, one recording/recordedsector holds 13 lines (12+1) of data. In FIG. 27A, "B0, 0, B0, 1, 2, . .. 15" designate the 16 recording/recorded sectors, respectively.

The video packs, sub picture packs and audio packs are interlaced on thetrack of the disk. However, this invention is not limited to thisarrangement of the packs. This invention can be applied to the diskwhich only the audio packs are arranged, or the disk which the audiopacks and sub packs are arranged, or the disk which the audio packs, subpacks and nav packs are arranged. It is free to combine the packs eachother.

Additional advantages and modifications will readily occur to thoseskilled in the art. Therefore, the invention in its broader aspects isnot limited to the specific details and representative embodiments shownand described herein. Accordingly, various modifications may be madewithout departing from the spirit or scope of the general inventiveconcept as defined by the appended claims and their equivalents.

We claim:
 1. A data arranging method for recording or transferring data,for use in a system for recording or transferring quantized dataobtained by sampling one channel or multi-channel signals in a timesequential manner and reproducing said quantized data, said methodcomprising the steps of:separating M-bit sample data of each channelsignal to a main word consisting of m1 bits on an MSB (Most SignificantBit) side and an extra word consisting of m2 bits on an LSB (LeastSignificant Bit) side; arranging a collection of main words of 2n-thsample data of individual channels as a main sample S2n; then arranginga collection of main words of (2n+1)-th sample data of individualchannels as a main sample S2n+1; then arranging a collection of extrawords of 2n-th sample data of individual channels as an extra samplee2n; and then arranging a collection of extra words of (2n+1)-th sampledata of individual channels as an extra sample e2n+1 (where n=0, 1, 2, .. . ), whereby resultant data is recorded on a recording medium ortransferred.
 2. A signal processing apparatus for recording ortransferring quantized data obtained by sampling one channel ormulti-channel signals in a time sequential manner and reproducing saidquantized data, said apparatus comprising:means for generating datahaving a structure in which M-bit sample data of each channel signal isseparated to a main word consisting of m1 bits on an MSB (MostSignificant Bit) side and an extra word consisting of m2 bits on an LSB(Least Significant Bit) side, a collection of main words of 2n-th sampledata of individual channels is arranged as a main sample S2n, acollection of main words of (2n+1)-th sample data of individual channelsis arranged next as a main sample S2n+1, a collection of extra words of2n-th sample data of individual channels is arranged as an extra samplee2n and a collection of extra words of (2n+1)-th sample data ofindividual channels is arranged as an extra sample e2n+1 (where n=0, 1,2, . . . ).
 3. A recording medium for processing and recording quantizeddata obtained by sampling one channel or multichannel signals in a timesequential manner and reproducing said quantized data, wherein recordedon said recording medium is data having a structure in which M-bitsample data of each channel signal is separated to a main wordconsisting of m1 bits on an MSB (Most Significant Bit) side and an extraword consisting of m2 bits on an LSB (Least Significant Bit) side, acollection of main words of 2n-th sample data of individual channels isarranged as a main sample S2n, a collection of main words of (2n+1)-thsample data of individual channels is arranged next as a main sampleS2n+1, a collection of extra words of 2n-th sample data of individualchannels is then arranged as an extra sample e2n and a collection ofextra words of (2n+1)-th sample data of individual channels is thenarranged as an extra sample e2n+1 (where n=0, 1, 2, . . . ).
 4. Therecording medium according to claim 3, wherein said extra word consistsof an integer multiple of 4 bits (4n bits; n=0, 1, 2, . . . ).
 5. Therecording medium according to claim 3, wherein said main word consistsof an integer multiple of 8 bits (8n bits; n=1, 2, 3, . . . ).
 6. Therecording medium according to claim 3, wherein said data consists of acollection of said main samples S21n, S2n+1 and said extra samples e2n,e2n+1 as a unit, a group is formed by a collection of a predeterminednumber of frames each formed by a collection of a predetermined numberof samples.
 7. The recording medium according to claim 3, wherein saiddata consists of a collection of said main samples S21n, S2n+1 and saidextra samples e2n, e2n+1 as a unit, each of frames is formed by acollection of a predetermined number of samples and is assigned to aplurality of audio packets which are arranged, mixed with video packetsand sub picture packets, between control packets.
 8. The recordingmedium according to claim 7, wherein each of said packets has apredetermined byte length; andwhen a plurality of main and extra samplesare arranged in said packet, a top of a first main sample is placed at apredetermined position of said packet, other samples are sequentiallyarranged after said first main sample, a total byte length of saidplurality of main and extra samples is equal to or smaller than amaximum byte length of said packet, and when said total byte length isless than said maximum byte length, invalid data of a stuffing byte or apadding byte is inserted in a remaining portion.
 9. The recording mediumaccording to claim 8, wherein said plurality of samples are linear PCMdata and said maximum byte length is 2013 bytes.
 10. The recordingmedium according to claim 8, wherein when said total byte length is lessthan said maximum byte length and said remaining portion has a length of7 bytes or less, said stuffing byte is inserted in a packet header, andwhen said total byte length is equal to or greater than said maximumbyte length, said padding byte is inserted at an end portion of saidpacket.
 11. The recording medium according to claim 8, wherein an evennumber of samples are arranged in one packet.
 12. The recording mediumaccording to claim 8, wherein said stuffing byte is given to a packetincluding a head of an audio frame, and said padding byte is given to apacket which does not include a head of an audio frame.
 13. A signalprocessing apparatus for, when processing quantized data obtained bysampling one channel or multi-channel signals in a time sequentialmanner and reproducing said quantized data, reproducing data recorded ona recording medium or transferred, said data having a structure in whichM-bit sample data of each channel signal is separated to a main wordconsisting of m1 bits on an MSB (Most Significant Bit) side and an extraword consisting of m2 bits on an LSB (Least Significant Bit) side, acollection of main words of 2n-th sample data of individual channels isarranged as a main sample S2n, a collection of main words of (2n+1)-thsample data of individual channels is then arranged next as a mainsample S2n+1, a collection of extra words of 2n-th sample data ofindividual channels is then arranged as an extra sample e2n and acollection of extra words of (2n+1)-th sample data of individualchannels is then arranged as an extra sample e2n+1 (where n=0, 1, 2, . .. ), said apparatus comprising:means for acquiring a reproduced outputof only a main word of at least said m1 bits in said data recorded onsaid recording medium or said transferred data.
 14. The signalprocessing apparatus according to claim 13, further comprising means ofcoupling a main word of said m1 bits of a predetermined channel and anextra word of said m2 bits of an associated channel, both on saidrecording medium, to acquire a reproduced output.
 15. The signalprocessing apparatus according to claim 13, wherein said m1 bits are 16bits, said m2 bits are 4 bits.
 16. The signal processing apparatusaccording to claim 13, wherein said m1 bits are 16 bits, said m2 bitsare 8 bits.
 17. The signal processing apparatus according to claim 13,wherein said data consists of a collection of said main samples S21n,S2n+1 and said extra samples e2n, e2n+1 as a unit, a group is formed bya collection of a predetermined number of frames each formed by acollection of a predetermined number of samples.
 18. The signalprocessing apparatus according to claim 13, wherein said data consistsof a collection of said main samples S21n, S2n+1 and said extra samplese2n, e2n+1 as a unit, each of frames is formed by a collection of apredetermined number of samples and is assigned to a plurality of audiopackets which are arranged, mixed with video packets and sub picturepackets, between control packets.
 19. The signal processing apparatusaccording to claim 18, further comprising:an input terminal for, when aplurality of main and extra samples are arranged in each of said packetseach having a predetermined byte length, receiving a sequence of packetshaving a top of a first main sample placed at a predetermined positionof said packet, other samples being sequentially arranged after saidfirst main sample, a total byte length of said plurality of main andextra samples being equal to or smaller than a maximum byte length ofsaid packet, invalid data of a stuffing byte or a padding byte beinginserted in a remaining portion when said total byte length is less thansaid maximum byte length; an input switch for dividing sample words ofsaid individual channel; a plurality of memories for storing said samplewords from said input switch; and a plurality of output switches forreading sample words of said individual channels from said memories ofsaid individual channels.
 20. The signal processing apparatus accordingto claim 19, wherein said input switch has distribution terminalsassociated with all channels included in samples in said packet.
 21. Thesignal processing apparatus according to claim 19, wherein saidplurality of main and extra samples are linear PCM data and said maximumbyte length is 2013 bytes.
 22. The signal processing apparatus accordingto claim 19, wherein when said total byte length is less than saidmaximum byte length and said remaining portion has a length of 7 bytesor less, said stuffing byte is inserted in a packet header, and whensaid total byte length is equal to or greater than said maximum bytelength, said padding byte is inserted at an end portion of said packet.23. A data arranging method for recording or transferring data, for usein a system for recording or transferring quantized data obtained bysampling one channel or multi-channel signals in a time sequentialmanner and reproducing said quantized data, said method comprising thesteps of:separating M-bit sample data of each channel signal to a mainword consisting of m1 bits on an MSB (Most Significant Bit) side and anextra word consisting of m2 bits on an LSB (Least Significant Bit) side;arranging a collection of main words of 2n-th sample data of individualchannels as a main sample S2n; then arranging a collection of main wordsof (2n+1-2k)-th sample data of individual channels as a main sampleS2n+1-2k; then arranging a collection of extra words of 2n-th sampledata of individual channels as an extra sample e2n; and then arranging acollection of extra words of (2n+1-2k)-th sample data of individualchannels as an extra sample e2n+1-2k (where n=0, 1, 2, . . . , and k=agiven integer), whereby resultant data is recorded on a recording mediumor transferred.
 24. A signal processing apparatus for use in a systemfor recording or transferring quantized data obtained by sampling onechannel or multi-channel signals in a time sequential manner andreproducing said quantized data, said apparatus comprising:means forgenerating data having a structure in which M-bit sample data of eachchannel signal is separated to a main word consisting of m1 bits on anMSB (Most Significant Bit) side and an extra word consisting of m2 bitson an LSB (Least Significant Bit) side, a collection of main words of2n-th sample data of individual channels is arranged as a main sampleS2n, a collection of main words of (2n+1-2k)-th sample data ofindividual channels is then arranged as a main sample S2n+1-2k, acollection of extra words of 2n-th sample data of individual channels isthen arranged as an extra sample e2n, and a collection of extra words of(2n+1-2k)-th sample data of individual channels is then arranged as anextra sample e2n+1-2k (where n=0, 1, 2, . . . , and k=a given integer),whereby resultant data is recorded on a recording medium or transferred.25. A recording medium for processing and recording quantized dataobtained by sampling one channel or multichannel signals in a timesequential manner and reproducing said quantized data, wherein recordedon said recording medium is data having a structure in which M-bitsample data of each channel signal is separated to a main wordconsisting of m1 bits on an MSB (Most Significant Bit) side and an extraword consisting of m2 bits on an LSB (Least Significant Bit) side, acollection of main words of 2n-th sample data of individual channels isarranged as a main sample S2n, a collection of main words of(2n+1-2k)-th sample data of individual channels is arranged next as amain sample S2n+1-2k, a collection of extra words of 2n-th sample dataof individual channels is then arranged as an extra sample e2n and acollection of extra words of (2n+1-2k)-th sample data of individualchannels is then arranged as an extra sample e2n+1-2k (where n=0, 1, 2,. . . , and k=a given integer).
 26. A signal processing apparatus for,when processing quantized data obtained by sampling one channel ormulti-channel signals in a time sequential manner and reproducing saidquantized data, reproducing data recorded on a recording medium ortransferred, said data having a structure in which M-bit sample data ofeach channel signal is separated to a main word consisting of m1 bits onan MSB (Most Significant Bit) side and an extra word consisting of m2bits on an LSB (Least Significant Bit) side, a collection of main wordsof 2n-th sample data of individual channels is arranged as a main sampleS2n, a collection of main words of (2n+1-2k)-th sample data ofindividual channels is then arranged next as a main sample S2n+1-2k, acollection of extra words of 2n-th sample data of individual channels isthen arranged as an extra sample e2n and a collection of extra words of(2n+1-2k)-th sample data of individual channels is then arranged as anextra sample e2n+1-2k (where n=0, 1, 2, . . . , and k=a given integer),said apparatus comprising:means for acquiring a reproduced output ofonly a main word of said m1 bits in said data recorded on said recordingmedium or said transferred data.
 27. A signal processing apparatus for,when processing quantized data obtained by sampling one channel ormulti-channel signals in a time sequential manner and reproducing saidquantized data, reproducing data recorded on a recording medium ortransferred, said data having a structure in which M-bit sample data ofeach channel signal is separated to a main word consisting of m1 bits onan MSB (Most Significant Bit) side and an extra word consisting of m2bits on an LSB (Least Significant Bit) side, a collection of main wordsof 2n-th sample data of individual channels is arranged as a main sampleS2n, a collection of main words of (2n+1-2k)-th sample data ofindividual channels is then arranged next as a main sample S2n+1-2k, acollection of extra words of 2n-th sample data of individual channels isthen arranged as an extra sample e2n and a collection of extra words of(2n+1-2k)-th sample data of individual channels is then arranged as anextra sample e2n+1-2k (where n=0, 1, 2, . . . , and k=a given integer),said apparatus comprising:means of coupling a main word of said m1 bitsof a predetermined channel and an extra word of said m2 bits of anassociated channel, both on said recording medium, to acquire areproduced output.